This reverts commit 30026214b13535a9fe1c47f1463378fbf976c643.
Reason for revert: Speculative revert, breaks downstream test.
Original change's description:
> Fix for out-of-bounds write in square test frame generator.
>
> The length is set on construction and includes an assumption on the
> image resolution, if the resolution changes, a square might be larger
> than what fits into the buffer, causing an out of bounds write. This
> CL fixes this simply by restricting the size of the square.
>
> Bug: webrtc:11415
> Change-Id: Iee14a1971997b4ae2fddef0a7af7c76a2509e879
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170042
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30732}
TBR=srte@webrtc.org,alito@webrtc.org
Change-Id: Ia0056da04a6f6f817ccadfc38aabe0c5f94754cc
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11415
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170115
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30743}
With the new config option |always_send_mid_and_rid|, the user
of the RTPSender can decide if MIDs and RIDs should always be sent
(when provided and negotiated), or if their sending should be disabled
after the receiver has acked the stream. Depending on the use case,
different settings might be preferable. The default setting is not
changed in this CL.
Bug: webrtc:11416
Change-Id: Ic3c71a6105fb7ee08d53cf9eb03f4e53b5799610
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170109
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30742}
The catapult code has learned how to deserialize protos, so we don't
need JSON as a middle step.
Also, set summary options to false for now to avoid polluting the
dashboard with _avg, _count for each metric. We don't use those
anyway. The durable solution is to set these from the histogram
writer, but catapult doesn't read the summary options correctly
from the proto yet.
Bug: chromium:1029452
Change-Id: I59d300fd34d36df836064ff41f6d0bf75bd6695e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170104
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30737}
This callback is enabled via the method
AudioCodingModule::RegisterVADCallback, which is unused, and deleted
in this cl.
Bug: None
Change-Id: I04c8690fbb673305e69fe5b1c32d88efd6c72d1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148420
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30735}
This reverts commit af1f8655b2cb69af382396ea642eb0a2bf04bb4d
Landing the change with default set to
"enabled" (DTLS 1.0 will continue to work by default),
so that flipping the default can be a separate CL.
Original change's description:
> Revert "Disable DTLS 1.0, TLS 1.0 and TLS 1.1 downgrade in WebRTC."
>
> This reverts commit 7276b974b78ea4f409d8738b1b6f1515f7a8968e.
>
> Reason for revert: Changing to a later Chrome release.
>
> Original change's description:
> > Disable DTLS 1.0, TLS 1.0 and TLS 1.1 downgrade in WebRTC.
> >
> > This change disables DTLS 1.0, TLS 1.0 and TLS 1.1 in WebRTC by default. This
> > is part of a larger effort at Google to remove old TLS protocols:
> > https://security.googleblog.com/2018/10/modernizing-transport-security.html
> >
> > For the M74 timeline I have added a disabled by default field trial
> > WebRTC-LegacyTlsProtocols which can be enabled to support these cipher suites
> > as consumers move away from these legacy cipher protocols but it will be off
> > in Chrome.
> >
> > This is compliant with the webrtc-security-arch specification which states:
> >
> > All Implementations MUST implement DTLS 1.2 with the
> > TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256 cipher suite and the P-256
> > curve [FIPS186]. Earlier drafts of this specification required DTLS
> > 1.0 with the cipher suite TLS_ECDHE_ECDSA_WITH_AES_128_CBC_SHA, and
> > at the time of this writing some implementations do not support DTLS
> > 1.2; endpoints which support only DTLS 1.2 might encounter
> > interoperability issues. The DTLS-SRTP protection profile
> > SRTP_AES128_CM_HMAC_SHA1_80 MUST be supported for SRTP.
> > Implementations MUST favor cipher suites which support (Perfect
> > Forward Secrecy) PFS over non-PFS cipher suites and SHOULD favor AEAD
> > over non-AEAD cipher suites.
> >
> > Bug: webrtc:10261
> > Change-Id: I847c567592911cc437f095376ad67585b4355fc0
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125141
> > Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> > Reviewed-by: David Benjamin <davidben@webrtc.org>
> > Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27006}
>
> TBR=steveanton@webrtc.org,davidben@webrtc.org,qingsi@webrtc.org,benwright@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:10261
> Change-Id: I34727e65c069e1fb2ad71838828ad0a22b5fe811
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130367
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Reviewed-by: Benjamin Wright <benwright@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27403}
Bug: webrtc:10261
Change-Id: I28c6819d37665976e396df280b4abf48fb91d533
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169851
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30733}
The length is set on construction and includes an assumption on the
image resolution, if the resolution changes, a square might be larger
than what fits into the buffer, causing an out of bounds write. This
CL fixes this simply by restricting the size of the square.
Bug: webrtc:11415
Change-Id: Iee14a1971997b4ae2fddef0a7af7c76a2509e879
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170042
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30732}
to avoid collission and confusion with VideoCodeType based on
c++ enum with the same name.
Bug: b/148146536
Change-Id: I049cce21d59f454c7ce507fdfc3a85d168f96223
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170048
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30728}
This is a reland of 49734dc0faa69616a58a1a95c7fc61a4610793cf
Patchset 2 contains a fix for the fuzzer set up. Since we now parse
an RtpPacket out of the fuzzer data, the header needs to be correct,
otherwise we fail before even reaching the FEC code that we actually
want to test.
Bug: webrtc:11340, chromium:1052323, chromium:1055974
TBR=stefan@webrtc.org
Original change's description:
> Reland "Refactors UlpFec and FlexFec to use a common interface."
>
> This is a reland of 11af1d7444fd7438766b7bc52cbd64752d72e32e
>
> Original change's description:
> > Refactors UlpFec and FlexFec to use a common interface.
> >
> > The new VideoFecGenerator is now injected into RtpSenderVideo,
> > and generalizes the usage.
> > This also prepares for being able to genera FEC in the RTP egress
> > module.
> >
> > Bug: webrtc:11340
> > Change-Id: I8aa873129b2fb4131eb3399ee88f6ea2747155a3
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168347
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30515}
>
> Bug: webrtc:11340, chromium:1052323
> Change-Id: Id646047365f1c46cca9e6f3e8eefa5151207b4a0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168608
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30593}
Bug: webrtc:11340, chromium:1052323
Change-Id: Ib8925f44e2edfcfeadc95c845c3bfc23822604ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169222
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30724}
This CL fixes the build for the meta taret "all"
(ninja -C out/Debug all).
More interestingly fixes cascaded_biquad_filter_unittest.cc which
seems not to be run at the moment.
Bug: webrtc:11411
Change-Id: I3d5f83c3898cca96aff8fbdad97d7b48caa9fffa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169858
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30722}
The EffectiveDegradationPreference() exposed an "implementation detail"
of the VideoStreamAdapter - how degradation preference may be modified.
By changing the return value of ApplyAdaptationTarget() this dependency
could be removed. We still have a TODO to get rid of the
ResourceListenerResponse enum, but that is QualityScaler related work.
This CL does the following:
- Module's GetAdaptUpTarget/GetAdaptDownTarget/ApplyAdaptationTarget
methods are removed in favor if invoking the VideoStreamAdapter's
version of these methods directly.
- Removing the EffectiveDegradationPreference() usage in
OveruseFrameDetectorResourceAdaptationModule meant moving that usage
to VideoStreamAdapter.
- MinPixelsPerFrame() is moved to VideoStreamAdapter; this is "can
adapt?" logic, i.e. the adapter's responsibility.
Bug: webrtc:11393
Change-Id: I75091ce97093bfa48a6d883492de30ed4b004492
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169859
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30714}
This makes an assumption that if we have variable frame length then we
will increase payload bitrate up to priority bitrate before adapting the
frame length.
Bug: webrtc:11001
Change-Id: Iec51d5ccce053d55ccd30a9e4712765227e10852
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169852
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30713}
This fixes two cases when the allocation is not updated correctly:
- The frame length range is not updated when audio network adaptor is enabled.
- The per-packet overhead is not updated unless the bitrate observer has been reconfigured.
Bug: webrtc:11001
Change-Id: I2ee25f956741a4be08661f874556582dd60a3bd0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169848
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30709}
This CL moves GetAdaptUpTarget(), GetAdaptDownTarget() and
ApplyAdaptationTarget() - and related code - to the VideoStreamAdapter.
This includes pieces related to calculating how to adapt, including:
- DegradationPreference
- BalancedDegradationPreference
- AdaptationRequest and last_adaptation_request_
- CanAdaptUpResolution()
The VideoStreamAdapter's interface has changed: VideoSourceRestrictor
methods are now hidden in favor of methods exposing AdaptationTarget.
This CL also does some misc moves:
- GetEncoderBitrateLimits is moved and renamed to
VideoEncoder::EncoderInfo::GetEncoderBitrateLimitsForResolution.
- EncoderSettings moved to a separate file.
// For api/video_codecs/video_encoder.[cc/h] changes, which is the
// moving of a function.
TBR=sprang@webrtc.org
Bug: webrtc:11393
Change-Id: Ie6bd8ef644ce927d7eca6ab90a0a7bcace682f3c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169842
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30708}
